Pass time constant to bwe smoothing filter.

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index a96eaf83f83a2928226651630e24a1b596c36587..23182021a18e5c5342c3745c8eca45167bdbf0ae 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -229,7 +229,8 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
-                                           int64_t rtt) {
+                                           int64_t rtt,
+                                           int64_t probing_interval_ms) {
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
@@ -238,7 +239,8 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
-  channel_proxy_->SetBitrate(bitrate_bps);
+  channel_proxy_->SetBitrate(bitrate_bps,
+                             rtc::Optional<int64_t>(probing_interval_ms));
 
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.